Signal processing method and arrangement for substitution or erroneous signals in a block-coded audio signals of an audio communication system

ABSTRACT

In order to improve the transmission quality of block-coded audio signals in audio communications systems, for example in DECT cordless telephones, when transmission errors occur, in such a manner that the requirement for computation power and thus the costs are minimal and no additional delay occurs to the audio signal to be transmitted, pauses in the audio signal which are caused by transmission errors are replaced by a pause-specific substitution signal which is generated in advance. In the simplest case, the substitution signal is generated by the signal section which immediately precedes a given signal section of the audio signal being buffer-stored and, if the given signal section is disturbed, being inserted into the gap which is produced by the disturbance.

BACKGROUND OF THE INVENTION

The invention relates to a signal processing method and apparatus forblock-coded audio signals of a communication system.

Transmitting and receiving devices are used for message processing andtransmission in communications systems having a message transmissionpath between a message origin and a message destination. The messageproduced by the message origin is transmitted by the transmitting devicevia a message channel to the receiving device, which subsequently emitsthe received message to the message destination. The message processingand transmission can in this case be carried out in a preferredtransmission direction or in both transmission directions (duplexoperation).

"Message" is a generic term which represents both the meaning content(information) and the physical representation (signal). Signals mayrepresent, for example,

(1) pictures,

(2) spoken words,

(3) written words,

(4) encrypted words or pictures.

The type of transmission according to (1), (2) and (3) is in this casenormally characterized by continuous (analog) signals, whilenon-continuous signals (for example pulses, digital signals) arenormally produced for the type of transmission according to (4).

The present invention primarily relates to the transmission of audiomessages (for example voice or music messages, etc.). However, it canalso be applied to other messages, such as appropriately processed videomessages, for example.

Either continuous signals (pure analog signals) or a mixture ofcontinuous and non-continuous signals occur as possible signal forms foran audio communications system, using A/D (analog to digital) convertersand D/A (digital to analog) converters. Devices which are specific tothe message type are in each case required for the functions of"transmitting" and "receiving". The question as to which of thesedevices is finally used also depends, inter alia, on the communicationschannel which is used as the basis for the audio communications system.The present invention thus primarily relates to telecommunicationssystems, which have a wire-free telecommunications channel.Telecommunications systems having such a structure are, for example,cordless telephones to the DECT standard (Digital European CordlessTelecommunication; cf. (1) European Telecommunication Standard; prETS300 175-1 . . . 9, October 1992, Parts 1 to 9, ETS-Institute 06921 SofiaAntipoles, France; (2) Nachrichtentechnik Elektronik 42(Telecommunications Electronics 42) (January/February 1992), No. 1,Berlin; U. Pilger: "Struktur des DECT-Standards" (Structure of the DECTStandard); pages 23 to 29; (3) Philips Telecommunication Review: "DECT,Universal Cordless Access System"; Vol. 49, No. 3, September 1991, pages68 to 73) or mobile radio telephones to the GSM standard (GroupeSpeciale Mobile Systems for Mobile Communication; cf. InformatikSpektrum (Information Spectrum), Springer Press Berlin, Year 14, 1991,No. 3, pages 137 to 152, "Der GSM-Standard--Grundlage fur digitaleeuropaische Mobilfunknetze" (The GSM Standard Basis for digital Europeanmobile radio networks)).

The DECT cordless telephone and the GSM mobile radio telephone are audiocommunications systems in which block-coded audio signals--for examplesignals which are coded using the TDMA or CDMA method (Time DivisionMultiple Access or Code Division Multiple Access)--are processed. Themessage transmitted using these telephones as a rule comprises,according to the above definition of message types, a mixture ofcontinuous and non-continuous signals. This signal mixture is in thiscase produced by the use of analog/digital and digital/analogconverters.

FIG. 1 shows a DECT cordless telephone having a cordless base station FT(Fixed Termination) to which a maximum of twelve cordless mobilesections (PT1 . . . PT12 (Portable Termination) are assigned forcordless telecommunication via a radio channel FK. Cordless basestations designed in such a way have been introduced to the market usingthe product name "Gigaset 952"--cf. DE-Z: the German journal FunkschauDecember 1993, pages 24 and 25; "Digitale Freiheit--Gigaset 952: Daserste DECT-Telefon" (Digital freedom--Gigaset 952: The first DECTtelephone); author: G. Weckwerth--1993. This design was essentially alsoknown before this from DE-Z: the German journal Funkschau October 1993;pages 74 to 77; title: "Digital kommunizieren mit DECT-DECT-Chipsatz vonPhilips" (Communicate digitally using the DECT-DECT chip set fromPhilips); author: Dr. J. Nieder and WO 94/10812 (FIG. 1 with theassociated description).

FIG. 2 shows the principle of the design of the DECT-specific cordlessmobile section PT1 . . . PT12, as is used for the transmission of voicemessages in the cordless telephone. Cordless mobile sections designed insuch a way have likewise been introduced to the market using the productname "Gigaset 952"--cf. DE-Z: the German journal Funkschau December1993, pages 24 and 25; "Digitale Freiheit--Gigaset 952: Das ersteDECT-Telefon" (Digital freedom--Gigaset 952: The first DECT telephone);author: G. Weckwerth--1993. This design was essentially also knownbefore this from DE-Z: the German journal Funkschau October 1993; pages74 to 77; title: "Digital kommunizieren mit DECT-DECT-Chipsatz vonPhilips" (Communicate digitally using the DECT-DECT chip set fromPhilips); author: Dr. J. Nieder and WO 94/10812 (FIG. 1 with theassociated description).

Block-oriented coding methods (for example TDMA methods) are used forthe transmission of voice and/or music signals (audio signals) with theDECT cordless telephone, in order on the one hand to use a correlationbetween signal sections which follow one another in time for datareduction and/or, on the other hand, to carry out block-oriented errorprotection by means of parity bits. If the transmission of the digitallycoded signals is disturbed, then bit errors obviously occur which can becompensated for, if the error rate is low, by the redundancy mechanismswhich are assigned to the digitally coded signal. However, if the biterror rates reach higher levels, an error correction is no longerpossible and an entire signal block will in consequence be identified asbeing faulty and will be rejected. There are a number of options at thereceiver end for coping with such signal blocks which have beenidentified as being faulty and have been rejected.

A first option, which is disclosed in WO 94/10769, comprises"squelching" the appropriate signal block which has been identified asbeing faulty, that is to say changing the code in an appropriate manner,for example by means of a sequence of zeros. This method is now used indigital DECT cordless telephones, such as Gigaset 952.

A second option for error correction is to assume that the error whichhas occurred is only minor. However, in this case, it is necessary todistinguish whether the algorithm can be used to identify the importanceof the respectively disturbed bits. In the case of conventional linearcoding, for example, a disturbed less significant bit (LSB=LeastSignificant Bit) would scarcely produce any audible errors, while anincorrectly set more significant bit (MSB=Most Significant Bit) wouldproduce severe sudden changes in the transmission signal and thuscrackling-like interference. Meanwhile, it is not possible in all casesto identify directly how severe the specific interference to be expectedwill be.

An entirely different way to correct errors in disturbed audio signalsis proposed in the documents:

(1) A. Papoulis: "A new algorithm in Spectral Analysis and Band-LimitedExtrapolation"; IEEE Transactions on Circuits and Systems, Volume 22(9), pages 735 ff., 1975 and

(2) R. Sottek: "Modelle zur Signalverarbeitung im menschlichen Gehor"[Models for signal processing in human hearing]; Thesis at the Institutefor Electrical Telecommunications, RWTH Aachen 1993.

A method is known from each of the cited documents, in which signalerrors in the audio signal which are caused by interference are maskedby interpolation of the signal. The disadvantages in the case of thismethod are, on the one hand, the high technical complexity which, undersome circumstances, demands the complete computation power of acurrently marketed digital signal processor (DSP=Digital SignalProcessor) and, on the other hand, makes the algorithmic delay of thesignal necessary, if processing is carried out in the frequency domainusing Fourier transformation. This delay would not be tolerable, forexample, in the case of telephony, particularly cordless telephony.

A method for the transmission of digital audio signals is disclosed inGerman reference DE-41 11 131 A1, in which a substitution signal whichis correlated with the signal is generated and buffer-stored forprocessing of the signals, at least one first incorrectly transmittedsignal section is determined in the signal, the first signal section ofthe signal is replaced by the substitution signal, andsubstitution-dependent artefacts in the signal are suppressed.

SUMMARY OF THE INVENTION

The object on which the invention is based is to improve thetransmission quality of block-coded audio signals in audiocommunications systems when transmission errors occur, in such a mannerthat the requirement for computation power and thus the costs areminimal, and, in particular, no additional delay or adverse effectoccurs to the audio signal to be transmitted.

In general terms the present invention is a signal processing method forblock-coded audio signals of a communications system. A substitutionsignal which is correlated with the audio signal is generated andbuffer-stored. At least one first, incorrectly transmitted signalsection is determined in the audio signal. The first signal section ofthe audio signal is replaced by the substitution signal.Substitution-dependent artefacts in the audio signal are suppressed. Afilter function is produced to suppress the artefacts, as a result ofwhich the substituion-dependent artefacts in the audio signal arefiltered such that the audio signal (on the basis of psycho-acousticaspects) is essentially maintained.

Advantageous developments of the present invention are as follows.

The audio signal is digitally filtered.

The audio signal is a voice signal.

The substitution signal is generated from a second, correctlytransmitted signal section of the audio signal, which is transmittedimmediately before the first signal section.

In the event of first signal sections which occur essentiallycontinuously, the suppression of the substitution-dependent artefacts ischanged in time. The audio signal is masked out if the first signalsections, which occur essentially continuously, exceed a predeterminedtime duration. The present invention is also an application of themethod as described above in a DECT-specific cordless telecommunicationssystem having at least one cordless base station and at least onecordless mobile section which is assigned to it.

The present invention is furthermore a signal processing arrangement forblock-coded audio signals of a communications system having a firstmeans for generation and buffer-storage of a substitution signal whichis correlated with the audio signal. A second means identifies at leastone first, incorrectly transmitted signal section in the audio signal. Athird means replaces the first signal section of the audio signal withthe substitution signal. A fourth means suppressessubstitution-dependent artefacts in the audio signal. The fourth meansare designed as a filter having a filter function for suppressing theartefacts, which filter filters the substitution-dependent artefacts inthe audio signal such that the audio signal (on the basis ofpsycho-acoustic aspects) is essentially maintained.

Advantageous developments of the present invention are as follows.

The fourth means is a digital filter.

The audio signal is a voice signal.

The first means are designed such that the substitution signal isgenerated from a second, correctly transmitted signal section of theaudio signal, which is transmitted immediately before the first signalsection. The second means, the third means and the fourth means form afunctional unit such that, in the event of first signal sections whichoccur essentially continuously, the suppression of thesubstitution-dependent artefacts is changed in time. The functional unitwhich is formed from the second to the fourth means is designed suchthat the audio is masked out if the first signal sections, which occuressentially continuously, exceed a predetermined time duration.

The first means is designed as a first program module, the second meansis designed as a second program module, the third means is designed as athird program module and the fourth means is designed as a fourthprogram module of a digital signal processor.

The digital filter is designed as a first-order recursive filter havinga low-pass filter characteristic.

The digital filter additionally has a high-pass element which suppressesartefacts which are produced by the repetition frequency in the event ofmultiple use of a common signal section for the substitution.

The signal processing arrangement as described above is used in at leastone cordless base station and in at least one cordless mobile section,which is assigned to the cordless base station, of a DECT-specificcordless telecommunications system.

The idea on which the invention is based is to replace the pauses in theaudio signal which are caused by transmission errors by a pause-specificsubstitution signal which is generated in advance.

In the simplest case, the substitution signal is generated by the signalsection which immediately precedes a given signal section of the audiosignal being buffer-stored and, if the given signal section isdisturbed, being inserted into the gap which is produced by thedisturbance. This procedure may even be used on its own since, in thecase of audio signals (music or voice signals), there is a high level ofcorrelation between signal sections which are closely adjacent to oneanother in time.

The reason for this is the fact that the mechanisms which produce volume(for example oscillation of chords in the case of music production,movements in the vocal tract in the case of voice production, etc.) havea certain amount of mechanical inertia. If signal sections of the audiosignal which follow one another in the order of magnitude of 10 to 20 msare compared, then a very high level of similarity is almost alwaysfound in the time signal (FIG. 3).

Alternatively, it is also possible to extend the generation of thepause-specific substitution signal initially over a plurality of signalsections which precede the given signal section of the audio signalsuccessively in time, and to buffer-store them, and then in the case ofa disturbed given signal section--to close the gap in the audio signal,which gap is caused by the disturbance, in the course of optimizedcontinuity matching which is carried out by comparison of the signalsection end of the last correctly transmitted signal section with thatstart of a substitution signal in the buffer-stored substitution signalwhich best matches this signal section end.

However, the replacement of the faulty signal section by precedingsignal sections using one of the methods described above leads (even inthe case of the method using optimized continuity matching) to theproblem that discontinuities can occur in the audio signal at theinsertion points because of the unknown phase of signal sections of theaudio signal (FIG. 4). The simple determination of a fundamentalfrequency of the audio signal in order, for example, continuously tomatch the signal section to be inserted to the preceding section isimpossible in the case of voice signals in telephony because this voicefundamental frequency--which is in the frequency spectrum between 160and 200 Hz--is filtered out by a high-pass filter (high-pass filteringat 300 Hz). On the other hand, it is possible to place signal sectionsalongside one another continuously only when the phases of theindividual frequency elements are also known, in addition to thespectral distribution. However, this in turn is dependent on continuousspectral analysis--for example using Fourier--which, however, isimpossible because of the already mentioned computational complexity.

The discontinuities mentioned above also lead to audible crackling inthe audio signal transmission being audible. A low-pass filter ispreferably used to suppress this high-frequency interference, forexample by suppressing higher spectral elements in the Fourier transformof the step function. The low-pass filter has a smoothing effect in thetime domain, while unnatural high-frequency elements are attenuated inthe frequency domain. The adverse affect caused by this to the audiosignal to be transmitted is tolerable if the low-pass filter, which ispreferably designed as a digital filter, does not chop the audio signaltoo severely. The tuning of the filter can thus be regarded as acompromise, which is optimized on a psycho-acoustic basis. Furthermore,it is a requirement that the filter can be switched on and off withoutdisturbances.

BRIEF DESCRIPTION OF THE DRAWINGS

The features of the present invention which are believed to be novel,are set forth with particularity in the appended claims. The invention,together with further objects and advantages, may best be understood byreference to the following description taken in conjunction with theaccompanying drawings, in the several Figures of which like referencenumerals identify like elements, and in which:

FIG. 1 depicts a DECT cordless telephone;

FIG. 2 depicts the principle of the design of the DECT--specificcordless mobile section;

FIG. 3 shows a voice signal for a spoken "a" split into a plurality oftime sections;

FIG. 4 shows the occurrence of discontinuities in the voice signal whentime sections are replaced (arrows);

FIG. 5 shows, based on FIG. 2, the modified topology of the cordlessmobile section PT in order to improve the transmission quality ofTDMA-specific (Time Division Multiple Access), DECT voice signals inDECT cordless telephones when transmission errors occur;

FIG. 6 shows the design of a simple filter for suppressingsubstitution-dependent artefacts in the voice signal;

FIG. 7 shows the measured transfer function of an actual low-pass filter(first-order recursive filter according to FIG. 6);

FIG. 8 shows a DECT voice signal which is disturbed over three (10 ms)periods;

FIG. 9 shows the DECT voice signal processed for the three (10 ms)periods according to FIG. 8.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 3 shows the time waveform of a voice signal SS_(a) for the spoken"a", which is split into a plurality of (10 ms) signal sections. Thesimilarity of adjacent time sections can be seen in this time divisionof the voice signal SS_(a). This correlation between sub-elements of thevoice signal SS_(a) which are close to one another is a result of thefact that the volume producing mechanisms (movements in the vocal tract)have a certain amount of mechanical inertia.

On the basis of the voice signal SS_(a) for the spoken "a" according toFIG. 3, FIG. 4 shows the same voice signal SS_(a) for a different timeaxis. In the case of the voice signal according to FIG. 4, a timesection in the time period between 40 and 50 ms has been replaced bycopying the preceding time section. This substitution has resulted indiscontinuities at the points marked by the arrows, which can be heardas crackling-like interference in the electro-acoustic signalconversion.

On the basis of FIG. 2, FIG. 5 shows the modified topology of thecordless mobile section PT for improving the transmission quality ofTDMA-specific (Time Division Multiple Access), DECT voice signals inDECT cordless telephones when transmission errors occur. Thetransmission errors frequently occur in the boundary regions when DECTcordless telephone radio messages are being transmitted, so thatDECT-specific burst and information losses occur in these regions.Because of this, the modified cordless mobile section PT has a digitalsignal processor DSP which is arranged on a transmission path US of thecordless mobile section PT--from an antenna ANT with a downstream radiosection FKT (transmitter/receiver) to an earpiece HK in the receivingdirection and from a microphone MIK to the antenna ANT in thetransmitting direction--between a signal control device BMC (Burst ModeController) and a signal conversion device SUE (Codec, AD/DA converter).The digital signal processor DSP is in this case controlled by afunction/sequence control device MIC (DECT microcontroller) which isspecific to the mobile section. In order that the digital signalprocessor DSP can improve the transmission quality of the DECT voicesignals, which are transmitted partially disturbed on the transmissionpath US, a plurality of program modules are assigned to the digitalsignal processor DSP,

(1) a first program module PM1 for generation and buffer-storage of asubstitution signal which is correlated with the voice signal,

(2) a second program module PM2 for determination of at least one first,incorrectly transmitted signal section in the voice signal,

(3) a third program module PM3 for replacement of the first signalsection of the voice signal by the substitution signal, and

(4) a fourth program module PM4 for suppression ofsubstitution-dependent artefacts in the DECT voice signal.

While the first three modules PM1, PM2, PM3 detect and evaluate the saidspecial features which are specific to the voice signal under thecontrol authority of the function/sequence control device MIC, thediscontinuities which occur in the DECT voice signal during theevaluation are filtered out digitally by the fourth program module PM4.In its preferred embodiment, the fourth program module PM4 is thereforejust a digital filter. The sudden discontinuities at the junction pointsof the time signal sections are smoothed out by the digital filter.

FIG. 6 shows the design of a digital filter which is implemented by theprogram module PM4 according to FIG. 5. In its simplest form, thisdigital filter is designed as a first-order recursive filter(IIR-Filter; Infinite Impulse Response-Filter). The recursive filter hasa filter function H(ω), which,

(1) at an angular frequency ω=0, has the function value H(ω=0)=b₀ *1/(1-a₁) and

(2) at an angular frequency ω=π, has the function value H(ω=π)=b₀ *1/(1+a₁).

In consequence, unique design of the filter is possible. If thecoefficient a₁ is in the value range between 0 and 1 (0<a₁ <1), then therecursive filter is a low-pass filter. Using the relationship b₀ =(1-a₁)and a value a₁ =0.8, the following filter function values result forω=0, ω=π/8 and ω=π: H(ω=0)=1, H(ω=π/8)=0.5 and H(ω=π)=0.111.

FIG. 7 shows the transfer function measured on an actual first-orderlow-pass filter. The resultant band cut-off at 4 kHz results from thebandwidth of the voice signal which is transmitted with theDECT-specific cordless mobile section PT at a sampling rate of 8 kHz.This filter produces a signal attenuation of just 20 dB at the highestfrequencies.

FIG. 8 shows a voice signal in which the DECT voice signal is disturbedover three (10 ms) periods (time period on the time axis between 4425 msand 4455 ms). The individual (10 ms) period corresponds to the timeduration of a TDMA time-division multiplex frame in the DECT cordlesssignal transmission.

If the signal which is illustrated in FIG. 8 is transmitted on thetransmission path US, which is illustrated in FIG. 5, of the cordlessmobile section PT, then the modified voice signal which is illustratedin FIG. 9 finally results at the output of the signal conversion deviceSUE according to FIG. 5. The difference which results in this case fromthe original voice signal according to FIG. 8 is the sole result of theprocessing of the original voice signal in the digital signal processorDSP according to FIGS. 5 and 6. The functional steps which are carriedout in the digital signal processor DSP on the basis of the programmodules PM1 . . . PM4 are, in this case:

(I) The determination of at least one incorrectly transmitted signalsection in the voice signal. With respect to the voice signal accordingto FIG. 8, these are the three disturbed (10 ms) signal sections.

(II) The buffer-storage of the last correctly transmitted signal sectionof the voice signal (generation of a substitution signal).

(III) The replacement of the three (10 ms) signal sections of theoriginal voice signal by the buffer-stored substitution signal.

(IV) The application of the filter function of the digital recursivefilter according to FIG. 6 to the modified voice signal producedaccording to steps (1) . . . (3).

The digital signal processor DSP requires only the last buffer-storedsample value for the last functional step--the calculation of the filterfunction. The two coefficients a₁, b₀ then just need to be converted.

If a number of signal sections are disturbed in the voice signal, as inthe case of the voice signal according to FIG. 8, then this error iscorrected by appropriate insertion of the last signal which wastransmitted without interference, at a plurality of times. This methodcan, of course, be used only to a limited extent--for (10 ms) signalsections, this limit is about (empirical values) a time duration of 50ms. It is pointless to use the method when error-free voice signals canno longer be received. Continuous repetition of the last disturbedsignal section would lead to an unnatural audible impression. If thelimit stated above is exceeded, then the described method is modifiedsuch that, after a number of disturbed signal sections have beenreplaced by the last correctly transmitted signal section, the voicesignal is subsequently masked out with a time constant of, for example,20 ms. This operation can be carried out by the digital signal processorDSP without any major computation complexity. Alternatively, it is alsopossible in the event of relatively long-lasting transmission errors todesign the digital filter to be variable with time. This can be done,for example, by the cut-off frequency of the filter being reduced andthe effect of the filter thus being enhanced. As a result of the digitalsignal processor DSP characteristics described above, this processor candistinguish how many signal sections (DECT bursts) have been transmittedincorrectly, and can accordingly react differently, depending on theduration of the disturbed signal section.

In the event of multiple repetition of one and the same voice signal, itis also possible for elements to be produced in the signal spectrumwhich correspond to the period of the signal section (time section) (forexample spectral elements of 100 Hz in the (10 ms) signal sections).These artefacts are partially attenuated by the high-pass response ofthe rest of the transmission path US of the cordless mobile section PTaccording to FIGS. 2 and 5 (for example by the frequency response in theear-piece HK). However, alternatively, it is also possible to provide ahigh-pass filter component in the digital filter as well. This high-passfilter characteristic filters out the low-frequency signal elements.This procedure further assists in making the signal which is being dealtwith--as hearing tests have shown--more realistic. In the case oftelephony, it is in any case known for the low frequencies, whichcorrespond to this frequency band, not to be transmitted.

The invention is not limited to the particular details of the method andapparatus depicted and other modifications and applications arecontemplated. Certain other changes may be made in the above describedmethod and apparatus without departing from the true spirit and scope ofthe invention herein involved. It is intended, therefore, that thesubject matter in the above depiction shall be interpreted asillustrative and not in a limiting sense.

What is claimed is:
 1. A signal processing method for block-coded audiosignals of a communications system, comprising the steps of:a)generating and buffer-storing a substitution signal which is correlatedwith the audio signal; b) determining at least one first, incorrectlytransmitted signal section in the audio signal; c) replacing one signalsections of the audio signal by the substitution signal for a calculatedreplacement durations d) producing a filter function; and e) suppressingsubstitution-dependent artefacts in the audio signal using the filterfunction, as a result of which the substitution-depended artefacts inthe audio signal are filtered such that the audio signal, based onpsycho-acoustic aspects, is substantially maintained, the filterfunction being effected by a time variant filter, in a digital signalprocessor which is adapted for smoothing sudden discontinuities thatresult from replacement of said one or more signal sections with saidsubstitution signal as a function of said replacement duration.
 2. Themethod as claimed in claim 1, wherein the audio signal is digitallyfiltered.
 3. The method as claimed in claim 1, wherein the audio signalis a voice signal.
 4. The method as claimed in claim 1, wherein thesubstitution signal is generated from a second, correctly transmittedsignal section of the audio signal, which is transmitted immediatelybefore the first signal section.
 5. The method as claimed in claim 1,wherein, if first signal sections occur substantially continuously, thesuppression of the substitution-dependent artefacts is changed in time.6. The method as claimed in claim 5, wherein the audio signal is maskedout if the first signal sections, which occur substantiallycontinuously, exceed a predetermined time duration.
 7. The method asclaimed in claim 1, wherein the method is used in a DECT (DigitalEuropean Cordless Telephone)-specific cordless telecommunications systemhaving at least one cordless base station and at least one cordlessmobile section which is assigned to the base station.
 8. The method asclaimed in claim 1, wherein said time variant filter can be switched onand off while producing no disturbances.
 9. A signal processingarrangement for block-coded audio signals of a communication system,comprising:a first system for generation and buffer-storage of asubstitution signal which is correlated with an audio signal; a secondsystem for identification of at least one first, incorrectly transmittedsignal section in the audio signal; a third system for replacement ofone or more signal sections of the audio signal with the substitutionsignal for a calculated replacement duration; a fourth system forsuppression of the substitution-depended artefacts in the audio signalsuch that the audio signal based on psycho-acoustic aspects, issubstantially maintained, said filter being a time variant filter, in adigital signal processor which is adapted for smoothing suddendiscontinuities that result from replacement of said one or more signalsections with said substitution signal as a function of said replacementduration.
 10. The signal processing arrangement as claimed in claim 9,wherein the audio signal is a voice signal.
 11. The signal processingarrangement as claimed in claim 9, wherein the first system effectsgeneration of the substitution signal from a second, correctlytransmitted signal section of the audio signal, which is transmittedimmediately before the first signal section.
 12. The signal processingarrangement as claimed in claim 9, wherein the second system, the thirdsystem and the fourth system form a functional unit such that, whenfirst signal sections occur, suppression of the substitution-dependentartefacts is changed in time.
 13. The signal processing arrangement asclaimed in claim 12, wherein the functional unit is structured such thatthe audio signal is masked out if the first signal sections, whichoccur, exceed a predetermined time duration.
 14. The signal processingarrangement as claimed in claim 9, wherein the arrangement is used in atleast one cordless base station and/or at least one cordless mobilesection, which is assigned to the cordless base station, of a DECT(Digital European Cordless Telephone)-specific cordlesstelecommunications system.
 15. The signal processing arrangement asclaimed in claim 9, wherein said time variant filter can be switched onand off while producing no disturbances.
 16. The signal processingarrangement as claimed in claim 9, wherein the fourth system forms adigital filter.
 17. The signal processing arrangement as claimed inclaim 16, wherein the arrangement further comprises a digital signalprocessor having first, second, third and fourth program modules andwherein the first system is the first program module, the second systemis the second program module, the third system is the third programmodule and the fourth system is the fourth program module of the digitalsignal processor.
 18. The signal processing arrangement as claimed inclaim 16, wherein the digital filter is a first-order recursive filterhaving a low-pass filter characteristic.
 19. The signal processingarrangement as claimed in claim 18, wherein the digital filteradditionally has a high-pass element which suppresses artefacts whichare produced by repetition frequency for multiple use of a common signalsection for the substitution.
 20. A signal processing method forblock-coded audio signals of a message system, comprising the stepsof:generating a substitution signal that is correlated with an audiosignal of said audio signals; temporarily storing said substitutionsignal; determining a first erroneously transmitted signal section insaid audio signal; replacing said first erroneously transmitted signalsection by said substitution signal, which thereby creates undesirablesubstitution-conditioned artefacts that arise at an insertion pointwhere said substitution signal is inserted into said audio signal due tophase differences between said replaced signal and said substitutionsignal; and suppressing said substitution-conditioned artefacts of saidaudio signal utilizing a filter function, wherein said filter functionsmooths said substitute-conditioned artefacts thereby maintainingpsycho-acoustic quality.
 21. A signal processing arrangement forblock-coded audio signals of a communication system, comprising:a firstsystem for generation and buffer-storage of a substitution signal whichis correlated with an audio signal of said audio signals; a secondsystem for identification of at least one first, incorrectly transmittedsignal section in the audio signal; a third system for replacement of afirst signal section of said audio signal with said substitution signalwhich thereby creates undesirable substitution-conditioned artefacts insaid audio signal that arise at an insertion point where saidsubstitution signal is inserted into said audio signal due to phasedifferences between a replaced signal and said substitution signal; afourth system for suppression of said substitution-conditionedartefacts, said fourth system having a filter that smooths saidsubstitute-conditioned artefacts thereby maintaining psycho-acousticquality.